Our last tutorial began a process of breaking down the broad range of network management functions into smaller pieces, based on the well-known Open Systems Interconnection (OSI) model. In that first part, we examined the Physical, Data Link and Network layers, which make up what is is sometimes called the Communications Subnetwork, and which are primarily implemented in hardware. Let’s continue that discussion, and look at the functions of the upper network layers of a VoIP networking environment.
Transport Layer: is responsible for the reliability of the end-to-end connection, and is typically implemented with two protocols. The Transmission Control Protocol (TCP) defined in RFC 793 (ftp://ftp.rfc-editor.org/in-notes/rfc793.txt ) provides the greatest amount of reliability, but at a cost of additional overhead. The User Datagram Protocol (UDP), defined in RFC 768 (ftp://ftp.rfc-editor.org/in-notes/rfc768.txt) cuts back on the reliability, but reduces the transmission overhead. The unit of information for TCP is called a stream, while the unit of information for UDP is called a datagram.
Most VoIP networks use TCP to setup the connection (due to its reliability), and then use UDP to actually transmit the VoIP packets (due to its efficiency). Thus, you pay the price of overhead when it really counts (establishing the connection), but subsequently cut back when the 20-to-40-millisecond voice packets start flowing, under the premise that if you miss a few voice packets along the way, the end users should still be able to make out the majority of the conversation.
Session Layer: provides for the establishment and termination of the communications sessions, and would be implemented as part of the H.323 standard from the International Telecommunication Union – Telecommunication Standardization Sector (ITU-T, see http://www.itu.int/ITU-T/), or the Session Initiation Protocol (SIP) functions developed by the Internet Engineering Task Force (IETF, see http://ietf.org/).
In the case of H.323, the call signaling functions bear some similarity to other WAN protocols, such as ISDN or ATM, as they were all developed from a common basis in legacy telephony. That part makes the network management of a VoIP network easier, as there is a lot of industry experience in diagnosing and troubleshooting WAN connections that go awry.
Unfortunately, another legacy of H.323 is complexity, which gave the IETF some of its incentive to develop SIP, which is less complex in the signaling area. But note the references above to protocols, and the need to use a protocol analyzer to sort out call establishment or disconnect procedures that do not operate as prescribed.
Presentation Layer: provides a mechanism of translating the sender’s data format into a format that could be understood at the receiver. In conventional networks, data encryption and data compression are good examples of Presentation layer functions. In the VoIP world, the codec (coder/decoder) that makes the analog-to-digital conversion provides an example of these capabilities.
But a problem can arise here: If your system selects one type of encoder (such as G.723.1) and I pick another (such as G.729), we may exchange bits, frames and packets, but will not be able to have an intelligible conversation (almost like speaking two foreign languages with no one available to translate). As before, this is likely to require a protocol analyzer to sort out.
Application Layer: provides functions that support the end user, such as voice dialing, unified messaging, or integration between landline, cellular, and VoIP systems. A failure here should be pretty obvious, since you might be able to complete the call using a dialing sequence, but the voice dialing, call forwarding, or follow-me application does not work. In most cases, these applications are embedded within an IP-based PBX or other switching system, and a call to the vendor would be a next logical step.
Speaking in generalities, the lower three OSI layers are typically implemented in hardware, and are therefore more accessible and straightforward to manage. The upper four layers are very likely implemented in software, and will require a protocol analyzer or more sophisticated system, such as a network management system based on the Simple Network Management Protocol (SNMP) to deal with these complexities.
Our next tutorial will examine the various tools that can be used for VoIP network management along with their capabilities.
Mark A. Miller, P.E., is President of DigiNet Corporation(R), a Denver-based consulting engineering firm. He is the author of many books on networking technologies, including Voice over IP Technologies, and Internet Technologies Handbook, both published by John Wiley & Sons.
Article courtesy of EnterpriseVoIPplanet.com, © DigiNet Corporation